On Sun, May 31, 2009 at 3:51 PM, sean darcy <[email protected]> wrote: > David Backeberg wrote: >> >> You don't say the kind of call you're making, but if you're using >> MeetMe() I have more advice regarding voice quality with conference >> rooms. >> > > I don't know about the OP, I'd sure appreciate any advice regarding > voice quality with MeetMe(). When we have 2 -3 internal SIP lines, 2+ > internet SIP lines, and some PRI lines, we have a difficult time with > quality. > > Any tips appreciated.
Sure. In addition to the things I mentioned, try jumping to the 1.6.1.* series. And be sure to NOT pass 'o' as an option to the conference. The 1.6.0. series had hard-coded talker optimization, which probably makes things nice for very heavily loaded conferences, but for our conferences was seeming to cause dropped voice packets that I assume were mistaken for line noise. We were able to reliably produce lost packets by making voice noises like breathing into the receiver, or moaning at the right pitch. In addition to those problems, it would clip the beginnings and endings of phrases. So if you were trying to tell somebody your phone number, like 555 555 5555, with breaks between each, you would have a very frustrating experience. You can read about a lengthy discussion on making optimization optional rather than mandatory at: https://issues.asterisk.org/view.php?id=13801 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
